WMAVoice decoder.

Originally committed as revision 21770 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 1568 insertions, 0 deletions

diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c
new file mode 100644
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+/*
+ * Windows Media Audio Voice decoder.
+ * Copyright (c) 2009 Ronald S. Bultje
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file libavcodec/wmavoice.c
+ * @brief Windows Media Audio Voice compatible decoder
+ * @author Ronald S. Bultje <rsbultje@gmail.com>
+ */
+
+#include <math.h>
+#include "avcodec.h"
+#include "get_bits.h"
+#include "put_bits.h"
+#include "wmavoice_data.h"
+#include "celp_math.h"
+#include "celp_filters.h"
+#include "acelp_vectors.h"
+#include "acelp_filters.h"
+#include "lsp.h"
+#include "libavutil/lzo.h"
+
+#define MAX_BLOCKS           8   ///< maximum number of blocks per frame
+#define MAX_LSPS             16  ///< maximum filter order
+#define MAX_FRAMES           3   ///< maximum number of frames per superframe
+#define MAX_FRAMESIZE        160 ///< maximum number of samples per frame
+#define MAX_SIGNAL_HISTORY   416 ///< maximum excitation signal history
+#define MAX_SFRAMESIZE       (MAX_FRAMESIZE * MAX_FRAMES)
+                                 ///< maximum number of samples per superframe
+#define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that
+                                 ///< was split over two packets
+#define VLC_NBITS            6   ///< number of bits to read per VLC iteration
+
+/**
+ * Frame type VLC coding.
+ */
+static VLC frame_type_vlc;
+
+/**
+ * Adaptive codebook types.
+ */
+enum {
+    ACB_TYPE_NONE       = 0, ///< no adaptive codebook (only hardcoded fixed)
+    ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which
+                             ///< we interpolate to get a per-sample pitch.
+                             ///< Signal is generated using an asymmetric sinc
+                             ///< window function
+                             ///< @note see #wmavoice_ipol1_coeffs
+    ACB_TYPE_HAMMING    = 2  ///< Per-block pitch with signal generation using
+                             ///< a Hamming sinc window function
+                             ///< @note see #wmavoice_ipol2_coeffs
+};
+
+/**
+ * Fixed codebook types.
+ */
+enum {
+    FCB_TYPE_SILENCE    = 0, ///< comfort noise during silence
+                             ///< generated from a hardcoded (fixed) codebook
+                             ///< with per-frame (low) gain values
+    FCB_TYPE_HARDCODED  = 1, ///< hardcoded (fixed) codebook with per-block
+                             ///< gain values
+    FCB_TYPE_AW_PULSES  = 2, ///< Pitch-adaptive window (AW) pulse signals,
+                             ///< used in particular for low-bitrate streams
+    FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in
+                             ///< combinations of either single pulses or
+                             ///< pulse pairs
+};
+
+/**
+ * Description of frame types.
+ */
+static const struct frame_type_desc {
+    uint8_t n_blocks;     ///< amount of blocks per frame (each block
+                          ///< (contains 160/#n_blocks samples)
+    uint8_t log_n_blocks; ///< log2(#n_blocks)
+    uint8_t acb_type;     ///< Adaptive codebook type (ACB_TYPE_*)
+    uint8_t fcb_type;     ///< Fixed codebook type (FCB_TYPE_*)
+    uint8_t dbl_pulses;   ///< how many pulse vectors have pulse pairs
+                          ///< (rather than just one single pulse)
+                          ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES
+    uint16_t frame_size;  ///< the amount of bits that make up the block
+                          ///< data (per frame)
+} frame_descs[17] = {
+    { 1, 0, ACB_TYPE_NONE,       FCB_TYPE_SILENCE,    0,   0 },
+    { 2, 1, ACB_TYPE_NONE,       FCB_TYPE_HARDCODED,  0,  28 },
+    { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES,  0,  46 },
+    { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2,  80 },
+    { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 },
+    { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 },
+    { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 },
+    { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 },
+    { 2, 1, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 0,  64 },
+    { 2, 1, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 2,  80 },
+    { 2, 1, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 5, 104 },
+    { 4, 2, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 0, 108 },
+    { 4, 2, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 2, 132 },
+    { 4, 2, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 5, 168 },
+    { 8, 3, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 0, 176 },
+    { 8, 3, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 2, 208 },
+    { 8, 3, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 5, 256 }
+};
+
+/**
+ * WMA Voice decoding context.
+ */
+typedef struct {
+    /**
+     * @defgroup struct_global Global values
+     * Global values, specified in the stream header / extradata or used
+     * all over.
+     * @{
+     */
+    GetBitContext gb;             ///< packet bitreader. During decoder init,
+                                  ///< it contains the extradata from the
+                                  ///< demuxer. During decoding, it contains
+                                  ///< packet data.
+    int8_t vbm_tree[25];          ///< converts VLC codes to frame type
+
+    int spillover_bitsize;        ///< number of bits used to specify
+                                  ///< #spillover_nbits in the packet header
+                                  ///< = ceil(log2(ctx->block_align << 3))
+    int history_nsamples;         ///< number of samples in history for signal
+                                  ///< prediction (through ACB)
+
+    int do_apf;                   ///< whether to apply the averaged
+                                  ///< projection filter (APF)
+
+    int lsps;                     ///< number of LSPs per frame [10 or 16]
+    int lsp_q_mode;               ///< defines quantizer defaults [0, 1]
+    int lsp_def_mode;             ///< defines different sets of LSP defaults
+                                  ///< [0, 1]
+    int frame_lsp_bitsize;        ///< size (in bits) of LSPs, when encoded
+                                  ///< per-frame (independent coding)
+    int sframe_lsp_bitsize;       ///< size (in bits) of LSPs, when encoded
+                                  ///< per superframe (residual coding)
+
+    int min_pitch_val;            ///< base value for pitch parsing code
+    int max_pitch_val;            ///< max value + 1 for pitch parsing
+    int pitch_nbits;              ///< number of bits used to specify the
+                                  ///< pitch value in the frame header
+    int block_pitch_nbits;        ///< number of bits used to specify the
+                                  ///< first block's pitch value
+    int block_pitch_range;        ///< range of the block pitch
+    int block_delta_pitch_nbits;  ///< number of bits used to specify the
+                                  ///< delta pitch between this and the last
+                                  ///< block's pitch value, used in all but
+                                  ///< first block
+    int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is
+                                  ///< from -this to +this-1)
+    uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale
+                                  ///< conversion
+
+    /**
+     * @}
+     * @defgroup struct_packet Packet values
+     * Packet values, specified in the packet header or related to a packet.
+     * A packet is considered to be a single unit of data provided to this
+     * decoder by the demuxer.
+     * @{
+     */
+    int spillover_nbits;          ///< number of bits of the previous packet's
+                                  ///< last superframe preceeding this
+                                  ///< packet's first full superframe (useful
+                                  ///< for re-synchronization also)
+    int has_residual_lsps;        ///< if set, superframes contain one set of
+                                  ///< LSPs that cover all frames, encoded as
+                                  ///< independent and residual LSPs; if not
+                                  ///< set, each frame contains its own, fully
+                                  ///< independent, LSPs
+    int skip_bits_next;           ///< number of bits to skip at the next call
+                                  ///< to #wmavoice_decode_packet() (since
+                                  ///< they're part of the previous superframe)
+
+    uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+                                  ///< cache for superframe data split over
+                                  ///< multiple packets
+    int sframe_cache_size;        ///< set to >0 if we have data from an
+                                  ///< (incomplete) superframe from a previous
+                                  ///< packet that spilled over in the current
+                                  ///< packet; specifies the amount of bits in
+                                  ///< #sframe_cache
+    PutBitContext pb;             ///< bitstream writer for #sframe_cache
+
+    /**
+     * @}
+     * @defgroup struct_frame Frame and superframe values
+     * Superframe and frame data - these can change from frame to frame,
+     * although some of them do in that case serve as a cache / history for
+     * the next frame or superframe.
+     * @{
+     */
+    double prev_lsps[MAX_LSPS];   ///< LSPs of the last frame of the previous
+                                  ///< superframe
+    int last_pitch_val;           ///< pitch value of the previous frame
+    int last_acb_type;            ///< frame type [0-2] of the previous frame
+    int pitch_diff_sh16;          ///< ((cur_pitch_val - #last_pitch_val)
+                                  ///< << 16) / #MAX_FRAMESIZE
+    float silence_gain;           ///< set for use in blocks if #ACB_TYPE_NONE
+
+    int aw_idx_is_ext;            ///< whether the AW index was encoded in
+                                  ///< 8 bits (instead of 6)
+    int aw_pulse_range;           ///< the range over which #aw_pulse_set1()
+                                  ///< can apply the pulse, relative to the
+                                  ///< value in aw_first_pulse_off. The exact
+                                  ///< position of the first AW-pulse is within
+                                  ///< [pulse_off, pulse_off + this], and
+                                  ///< depends on bitstream values; [16 or 24]
+    int aw_n_pulses[2];           ///< number of AW-pulses in each block; note
+                                  ///< that this number can be negative (in
+                                  ///< which case it basically means "zero")
+    int aw_first_pulse_off[2];    ///< index of first sample to which to
+                                  ///< apply AW-pulses, or -0xff if unset
+    int aw_next_pulse_off_cache;  ///< the position (relative to start of the
+                                  ///< second block) at which pulses should
+                                  ///< start to be positioned, serves as a
+                                  ///< cache for pitch-adaptive window pulses
+                                  ///< between blocks
+
+    int frame_cntr;               ///< current frame index [0 - 0xFFFE]; is
+                                  ///< only used for comfort noise in #pRNG()
+    float gain_pred_err[6];       ///< cache for gain prediction
+    float excitation_history[MAX_SIGNAL_HISTORY];
+                                  ///< cache of the signal of previous
+                                  ///< superframes, used as a history for
+                                  ///< signal generation
+    float synth_history[MAX_LSPS]; ///< see #excitation_history
+    /**
+     * @}
+     */
+} WMAVoiceContext;
+
+/**
+ * Sets up the variable bit mode (VBM) tree from container extradata.
+ * @param gb bit I/O context.
+ *           The bit context (s->gb) should be loaded with byte 23-46 of the
+ *           container extradata (i.e. the ones containing the VBM tree).
+ * @param vbm_tree pointer to array to which the decoded VBM tree will be
+ *                 written.
+ * @return 0 on success, <0 on error.
+ */
+static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
+{
+    static const uint8_t bits[] = {
+         2,  2,  2,  4,  4,  4,
+         6,  6,  6,  8,  8,  8,
+        10, 10, 10, 12, 12, 12,
+        14, 14, 14, 14
+    };
+    static const uint16_t codes[] = {
+          0x0000, 0x0001, 0x0002,        //              00/01/10
+          0x000c, 0x000d, 0x000e,        //           11+00/01/10
+          0x003c, 0x003d, 0x003e,        //         1111+00/01/10
+          0x00fc, 0x00fd, 0x00fe,        //       111111+00/01/10
+          0x03fc, 0x03fd, 0x03fe,        //     11111111+00/01/10
+          0x0ffc, 0x0ffd, 0x0ffe,        //   1111111111+00/01/10
+          0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
+    };
+    int cntr[8], n, res;
+
+    memset(vbm_tree, 0xff, sizeof(vbm_tree));
+    memset(cntr,     0,    sizeof(cntr));
+    for (n = 0; n < 17; n++) {
+        res = get_bits(gb, 3);
+        if (cntr[res] > 3) // should be >= 3 + (res == 7))
+            return -1;
+        vbm_tree[res * 3 + cntr[res]++] = n;
+    }
+    INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
+                    bits, 1, 1, codes, 2, 2, 132);
+    return 0;
+}
+
+/**
+ * Set up decoder with parameters from demuxer (extradata etc.).
+ */
+static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
+{
+    int n, flags, pitch_range, lsp16_flag;
+    WMAVoiceContext *s = ctx->priv_data;
+
+    /**
+     * Extradata layout:
+     * - byte  0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
+     * - byte 19-22: flags field (annoyingly in LE; see below for known
+     *               values),
+     * - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
+     *               rest is 0).
+     */
+    if (ctx->extradata_size != 46) {
+        av_log(ctx, AV_LOG_ERROR,
+               "Invalid extradata size %d (should be 46)\n",
+               ctx->extradata_size);
+        return -1;
+    }
+    flags                = AV_RL32(ctx->extradata + 18);
+    s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
+    s->do_apf            =    flags & 0x1;
+    s->lsp_q_mode        = !!(flags & 0x2000);
+    s->lsp_def_mode      = !!(flags & 0x4000);
+    lsp16_flag           =    flags & 0x1000;
+    if (lsp16_flag) {
+        s->lsps               = 16;
+        s->frame_lsp_bitsize  = 34;
+        s->sframe_lsp_bitsize = 60;
+    } else {
+        s->lsps               = 10;
+        s->frame_lsp_bitsize  = 24;
+        s->sframe_lsp_bitsize = 48;
+    }
+    for (n = 0; n < s->lsps; n++)
+        s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
+
+    init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3);
+    if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) {
+        av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
+        return -1;
+    }
+
+    s->min_pitch_val    = ((ctx->sample_rate << 8)      /  400 + 50) >> 8;
+    s->max_pitch_val    = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8;
+    pitch_range         = s->max_pitch_val - s->min_pitch_val;
+    s->pitch_nbits      = av_ceil_log2(pitch_range);
+    s->last_pitch_val   = 40;
+    s->last_acb_type    = ACB_TYPE_NONE;
+    s->history_nsamples = s->max_pitch_val + 8;
+
+    if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) {
+        int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
+            max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8;
+
+        av_log(ctx, AV_LOG_ERROR,
+               "Unsupported samplerate %d (min=%d, max=%d)\n",
+               ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
+
+        return -1;
+    }
+
+    s->block_conv_table[0]      = s->min_pitch_val;
+    s->block_conv_table[1]      = (pitch_range * 25) >> 6;
+    s->block_conv_table[2]      = (pitch_range * 44) >> 6;
+    s->block_conv_table[3]      = s->max_pitch_val - 1;
+    s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
+    s->block_delta_pitch_nbits  = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
+    s->block_pitch_range        = s->block_conv_table[2] +
+                                  s->block_conv_table[3] + 1 +
+                                  2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
+    s->block_pitch_nbits        = av_ceil_log2(s->block_pitch_range);
+
+    ctx->sample_fmt             = SAMPLE_FMT_FLT;
+
+    return 0;
+}
+
+/**
+ * Dequantize LSPs
+ * @param lsps output pointer to the array that will hold the LSPs
+ * @param num number of LSPs to be dequantized
+ * @param values quantized values, contains n_stages values
+ * @param sizes range (i.e. max value) of each quantized value
+ * @param n_stages number of dequantization runs
+ * @param table dequantization table to be used
+ * @param mul_q LSF multiplier
+ * @param base_q base (lowest) LSF values
+ */
+static void dequant_lsps(double *lsps, int num,
+                         const uint16_t *values,
+                         const uint16_t *sizes,
+                         int n_stages, const uint8_t *table,
+                         const double *mul_q,
+                         const double *base_q)
+{
+    int n, m;
+
+    memset(lsps, 0, num * sizeof(*lsps));
+    for (n = 0; n < n_stages; n++) {
+        const uint8_t *t_off = &table[values[n] * num];
+        double base = base_q[n], mul = mul_q[n];
+
+        for (m = 0; m < num; m++)
+            lsps[m] += base + mul * t_off[m];
+
+        table += sizes[n] * num;
+    }
+}
+
+/**
+ * @defgroup lsp_dequant LSP dequantization routines
+ * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
+ * @note we assume enough bits are available, caller should check.
+ * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
+ * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
+ * @{
+ */
+/**
+ * Parse 10 independently-coded LSPs.
+ */
+static void dequant_lsp10i(GetBitContext *gb, double *lsps)
+{
+    static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
+    static const double mul_lsf[4] = {
+        5.2187144800e-3,    1.4626986422e-3,
+        9.6179549166e-4,    1.1325736225e-3
+    };
+    static const double base_lsf[4] = {
+        M_PI * -2.15522e-1, M_PI * -6.1646e-2,
+        M_PI * -3.3486e-2,  M_PI * -5.7408e-2
+    };
+    uint16_t v[4];
+
+    v[0] = get_bits(gb, 8);
+    v[1] = get_bits(gb, 6);
+    v[2] = get_bits(gb, 5);
+    v[3] = get_bits(gb, 5);
+
+    dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
+                 mul_lsf, base_lsf);
+}
+
+/**
+ * Parse 10 independently-coded LSPs, and then derive the tables to
+ * generate LSPs for the other frames from them (residual coding).
+ */
+static void dequant_lsp10r(GetBitContext *gb,
+                           double *i_lsps, const double *old,
+                           double *a1, double *a2, int q_mode)
+{
+    static const uint16_t vec_sizes[3] = { 128, 64, 64 };
+    static const double mul_lsf[3] = {
+        2.5807601174e-3,    1.2354460219e-3,   1.1763821673e-3
+    };
+    static const double base_lsf[3] = {
+        M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
+    };
+    const float (*ipol_tab)[2][10] = q_mode ?
+        wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
+    uint16_t interpol, v[3];
+    int n;
+
+    dequant_lsp10i(gb, i_lsps);
+
+    interpol = get_bits(gb, 5);
+    v[0]     = get_bits(gb, 7);
+    v[1]     = get_bits(gb, 6);
+    v[2]     = get_bits(gb, 6);
+
+    for (n = 0; n < 10; n++) {
+        double delta = old[n] - i_lsps[n];
+        a1[n]        = ipol_tab[interpol][0][n] * delta + i_lsps[n];
+        a1[10 + n]   = ipol_tab[interpol][1][n] * delta + i_lsps[n];
+    }
+
+    dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
+                 mul_lsf, base_lsf);
+}
+
+/**
+ * Parse 16 independently-coded LSPs.
+ */
+static void dequant_lsp16i(GetBitContext *gb, double *lsps)
+{
+    static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
+    static const double mul_lsf[5] = {
+        3.3439586280e-3,    6.9908173703e-4,
+        3.3216608306e-3,    1.0334960326e-3,
+        3.1899104283e-3
+    };
+    static const double base_lsf[5] = {
+        M_PI * -1.27576e-1, M_PI * -2.4292e-2,
+        M_PI * -1.28094e-1, M_PI * -3.2128e-2,
+        M_PI * -1.29816e-1
+    };
+    uint16_t v[5];
+
+    v[0] = get_bits(gb, 8);
+    v[1] = get_bits(gb, 6);
+    v[2] = get_bits(gb, 7);
+    v[3] = get_bits(gb, 6);
+    v[4] = get_bits(gb, 7);
+
+    dequant_lsps( lsps,     5,  v,     vec_sizes,    2,
+                 wmavoice_dq_lsp16i1,  mul_lsf,     base_lsf);
+    dequant_lsps(&lsps[5],  5, &v[2], &vec_sizes[2], 2,
+                 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
+    dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
+                 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
+}
+
+/**
+ * Parse 16 independently-coded LSPs, and then derive the tables to
+ * generate LSPs for the other frames from them (residual coding).
+ */
+static void dequant_lsp16r(GetBitContext *gb,
+                           double *i_lsps, const double *old,
+                           double *a1, double *a2, int q_mode)
+{
+    static const uint16_t vec_sizes[3] = { 128, 128, 128 };
+    static const double mul_lsf[3] = {
+        1.2232979501e-3,   1.4062241527e-3,   1.6114744851e-3
+    };
+    static const double base_lsf[3] = {
+        M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
+    };
+    const float (*ipol_tab)[2][16] = q_mode ?
+        wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
+    uint16_t interpol, v[3];
+    int n;
+
+    dequant_lsp16i(gb, i_lsps);
+
+    interpol = get_bits(gb, 5);
+    v[0]     = get_bits(gb, 7);
+    v[1]     = get_bits(gb, 7);
+    v[2]     = get_bits(gb, 7);
+
+    for (n = 0; n < 16; n++) {
+        double delta = old[n] - i_lsps[n];
+        a1[n]        = ipol_tab[interpol][0][n] * delta + i_lsps[n];
+        a1[16 + n]   = ipol_tab[interpol][1][n] * delta + i_lsps[n];
+    }
+
+    dequant_lsps( a2,     10,  v,     vec_sizes,    1,
+                 wmavoice_dq_lsp16r1,  mul_lsf,     base_lsf);
+    dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
+                 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
+    dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
+                 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
+}
+
+/**
+ * @}
+ * @defgroup aw Pitch-adaptive window coding functions
+ * The next few functions are for pitch-adaptive window coding.
+ * @{
+ */
+/**
+ * Parse the offset of the first pitch-adaptive window pulses, and
+ * the distribution of pulses between the two blocks in this frame.
+ * @param s WMA Voice decoding context private data
+ * @param gb bit I/O context
+ * @param pitch pitch for each block in this frame
+ */
+static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
+                            const int *pitch)
+{
+    static const int16_t start_offset[94] = {
+        -11,  -9,  -7,  -5,  -3,  -1,   1,   3,   5,   7,   9,  11,
+         13,  15,  18,  17,  19,  20,  21,  22,  23,  24,  25,  26,
+         27,  28,  29,  30,  31,  32,  33,  35,  37,  39,  41,  43,
+         45,  47,  49,  51,  53,  55,  57,  59,  61,  63,  65,  67,
+         69,  71,  73,  75,  77,  79,  81,  83,  85,  87,  89,  91,
+         93,  95,  97,  99, 101, 103, 105, 107, 109, 111, 113, 115,
+        117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
+        141, 143, 145, 147, 149, 151, 153, 155, 157, 159
+    };
+    int bits, offset;
+
+    /* position of pulse */
+    s->aw_idx_is_ext = 0;
+    if ((bits = get_bits(gb, 6)) >= 54) {
+        s->aw_idx_is_ext = 1;
+        bits += (bits - 54) * 3 + get_bits(gb, 2);
+    }
+
+    /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
+     * the distribution of the pulses in each block contained in this frame. */
+    s->aw_pulse_range        = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
+    for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
+    s->aw_n_pulses[0]        = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
+    s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
+    offset                  += s->aw_n_pulses[0] * pitch[0];
+    s->aw_n_pulses[1]        = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
+    s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
+
+    /* if continuing from a position before the block, reset position to
+     * start of block (when corrected for the range over which it can be
+     * spread in aw_pulse_set1()). */
+    if (start_offset[bits] < MAX_FRAMESIZE / 2) {
+        while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
+            s->aw_first_pulse_off[1] -= pitch[1];
+        if (start_offset[bits] < 0)
+            while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
+                s->aw_first_pulse_off[0] -= pitch[0];
+    }
+}
+
+/**
+ * Apply second set of pitch-adaptive window pulses.
+ * @param s WMA Voice decoding context private data
+ * @param gb bit I/O context
+ * @param block_idx block index in frame [0, 1]
+ * @param fcb structure containing fixed codebook vector info
+ */
+static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
+                          int block_idx, AMRFixed *fcb)
+{
+    uint16_t use_mask[7]; // only 5 are used, rest is padding
+    /* in this function, idx is the index in the 80-bit (+ padding) use_mask
+     * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
+     * of idx are the position of the bit within a particular item in the
+     * array (0 being the most significant bit, and 15 being the least
+     * significant bit), and the remainder (>> 4) is the index in the
+     * use_mask[]-array. This is faster and uses less memory than using a
+     * 80-byte/80-int array. */
+    int pulse_off = s->aw_first_pulse_off[block_idx],
+        pulse_start, n, idx, range, aidx, start_off = 0;
+
+    /* set offset of first pulse to within this block */
+    if (s->aw_n_pulses[block_idx] > 0)
+        while (pulse_off + s->aw_pulse_range < 1)
+            pulse_off += fcb->pitch_lag;
+
+    /* find range per pulse */
+    if (s->aw_n_pulses[0] > 0) {
+        if (block_idx == 0) {
+            range = 32;
+        } else /* block_idx = 1 */ {
+            range = 8;
+            if (s->aw_n_pulses[block_idx] > 0)
+                pulse_off = s->aw_next_pulse_off_cache;
+        }
+    } else
+        range = 16;
+    pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
+
+    /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
+     * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
+     * we exclude that range from being pulsed again in this function. */
+    memset( use_mask,   -1, 5 * sizeof(use_mask[0]));
+    memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
+    if (s->aw_n_pulses[block_idx] > 0)
+        for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) {
+            int excl_range         = s->aw_pulse_range; // always 16 or 24
+            uint16_t *use_mask_ptr = &use_mask[idx >> 4];
+            int first_sh           = 16 - (idx & 15);
+            *use_mask_ptr++       &= 0xFFFF << first_sh;
+            excl_range            -= first_sh;
+            if (excl_range >= 16) {
+                *use_mask_ptr++    = 0;
+                *use_mask_ptr     &= 0xFFFF >> (excl_range - 16);
+            } else
+                *use_mask_ptr     &= 0xFFFF >> excl_range;
+        }
+
+    /* find the 'aidx'th offset that is not excluded */
+    aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4);
+    for (n = 0; n <= aidx; pulse_start++) {
+        for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ;
+        if (idx >= MAX_FRAMESIZE / 2) { // find from zero
+            if (use_mask[0])      idx = 0x0F;
+            else if (use_mask[1]) idx = 0x1F;
+            else if (use_mask[2]) idx = 0x2F;
+            else if (use_mask[3]) idx = 0x3F;
+            else if (use_mask[4]) idx = 0x4F;
+            else                  return;
+            idx -= av_log2_16bit(use_mask[idx >> 4]);
+        }
+        if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
+            use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
+            n++;
+            start_off = idx;
+        }
+    }
+
+    fcb->x[fcb->n] = start_off;
+    fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0;
+    fcb->n++;
+
+    /* set offset for next block, relative to start of that block */
+    n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
+    s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
+}
+
+/**
+ * Apply first set of pitch-adaptive window pulses.
+ * @param s WMA Voice decoding context private data
+ * @param gb bit I/O context
+ * @param block_idx block index in frame [0, 1]
+ * @param fcb storage location for fixed codebook pulse info
+ */
+static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
+                          int block_idx, AMRFixed *fcb)
+{
+    int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
+    float v;
+
+    if (s->aw_n_pulses[block_idx] > 0) {
+        int n, v_mask, i_mask, sh, n_pulses;
+
+        if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
+            n_pulses = 3;
+            v_mask   = 8;
+            i_mask   = 7;
+            sh       = 4;
+        } else { // 4 pulses, 1:sign + 2:index each
+            n_pulses = 4;
+            v_mask   = 4;
+            i_mask   = 3;
+            sh       = 3;
+        }
+
+        for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
+            fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
+            fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
+                                 s->aw_first_pulse_off[block_idx];
+            while (fcb->x[fcb->n] < 0)
+                fcb->x[fcb->n] += fcb->pitch_lag;
+            if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
+                fcb->n++;
+        }
+    } else {
+        int num2 = (val & 0x1FF) >> 1, delta, idx;
+
+        if (num2 < 1 * 79)      { delta = 1; idx = num2 + 1; }
+        else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
+        else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
+        else                    { delta = 7; idx = num2 + 1 - 3 * 75; }
+        v = (val & 0x200) ? -1.0 : 1.0;
+
+        fcb->no_repeat_mask |= 3 << fcb->n;
+        fcb->x[fcb->n]       = idx - delta;
+        fcb->y[fcb->n]       = v;
+        fcb->x[fcb->n + 1]   = idx;
+        fcb->y[fcb->n + 1]   = (val & 1) ? -v : v;
+        fcb->n              += 2;
+    }
+}
+
+/**
+ * @}
+ *
+ * Generate a random number from frame_cntr and block_idx, which will lief
+ * in the range [0, 1000 - block_size] (so it can be used as an index in a
+ * table of size 1000 of which you want to read block_size entries).
+ *
+ * @param frame_cntr current frame number
+ * @param block_num current block index
+ * @param block_size amount of entries we want to read from a table
+ *                   that has 1000 entries
+ * @returns a (non-)random number in the [0, 1000 - block_size] range.
+ */
+static int pRNG(int frame_cntr, int block_num, int block_size)
+{
+    /* array to simplify the calculation of z:
+     * y = (x % 9) * 5 + 6;
+     * z = (49995 * x) / y;
+     * Since y only has 9 values, we can remove the division by using a
+     * LUT and using FASTDIV-style divisions. For each of the 9 values
+     * of y, we can rewrite z as:
+     * z = x * (49995 / y) + x * ((49995 % y) / y)
+     * In this table, each col represents one possible value of y, the
+     * first number is 49995 / y, and the second is the FASTDIV variant
+     * of 49995 % y / y. */
+    static const unsigned int div_tbl[9][2] = {
+        { 8332,  3 * 715827883U }, // y =  6
+        { 4545,  0 * 390451573U }, // y = 11
+        { 3124, 11 * 268435456U }, // y = 16
+        { 2380, 15 * 204522253U }, // y = 21
+        { 1922, 23 * 165191050U }, // y = 26
+        { 1612, 23 * 138547333U }, // y = 31
+        { 1388, 27 * 119304648U }, // y = 36
+        { 1219, 16 * 104755300U }, // y = 41
+        { 1086, 39 *  93368855U }  // y = 46
+    };
+    unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
+    if (x >= 0xFFFF) x -= 0xFFFF;   // max value of x is 8*1877+0xFFFE=0x13AA6,
+                                    // so this is effectively a modulo (%)
+    y = x - 9 * MULH(477218589, x); // x % 9
+    z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
+                                    // z = x * 49995 / (y * 5 + 6)
+    return z % (1000 - block_size);
+}
+
+/**
+ * Parse hardcoded signal for a single block.
+ * @note see #synth_block().
+ */
+static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
+                                 int block_idx, int size,
+                                 const struct frame_type_desc *frame_desc,
+                                 float *excitation)
+{
+    float gain;
+    int n, r_idx;
+
+    assert(size <= MAX_FRAMESIZE);
+
+    /* Set the offset from which we start reading wmavoice_std_codebook */
+    if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
+        r_idx = pRNG(s->frame_cntr, block_idx, size);
+        gain  = s->silence_gain;
+    } else /* FCB_TYPE_HARDCODED */ {
+        r_idx = get_bits(gb, 8);
+        gain  = wmavoice_gain_universal[get_bits(gb, 6)];
+    }
+
+    /* Clear gain prediction parameters */
+    memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
+
+    /* Apply gain to hardcoded codebook and use that as excitation signal */
+    for (n = 0; n < size; n++)
+        excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
+}
+
+/**
+ * Parse FCB/ACB signal for a single block.
+ * @note see #synth_block().
+ */
+static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
+                                int block_idx, int size,
+                                int block_pitch_sh2,
+                                const struct frame_type_desc *frame_desc,
+                                float *excitation)
+{
+    static const float gain_coeff[6] = {
+        0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
+    };
+    float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
+    int n, idx, gain_weight;
+    AMRFixed fcb;
+
+    assert(size <= MAX_FRAMESIZE / 2);
+    memset(pulses, 0, sizeof(*pulses) * size);
+
+    fcb.pitch_lag      = block_pitch_sh2 >> 2;
+    fcb.pitch_fac      = 1.0;
+    fcb.no_repeat_mask = 0;
+    fcb.n              = 0;
+
+    /* For the other frame types, this is where we apply the innovation
+     * (fixed) codebook pulses of the speech signal. */
+    if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
+        aw_pulse_set1(s, gb, block_idx, &fcb);
+        aw_pulse_set2(s, gb, block_idx, &fcb);
+    } else /* FCB_TYPE_EXC_PULSES */ {
+        int offset_nbits = 5 - frame_desc->log_n_blocks;
+
+        fcb.no_repeat_mask = -1;
+        /* similar to ff_decode_10_pulses_35bits(), but with single pulses
+         * (instead of double) for a subset of pulses */
+        for (n = 0; n < 5; n++) {
+            float sign;
+            int pos1, pos2;
+
+            sign           = get_bits1(gb) ? 1.0 : -1.0;
+            pos1           = get_bits(gb, offset_nbits);
+            fcb.x[fcb.n]   = n + 5 * pos1;
+            fcb.y[fcb.n++] = sign;
+            if (n < frame_desc->dbl_pulses) {
+                pos2           = get_bits(gb, offset_nbits);
+                fcb.x[fcb.n]   = n + 5 * pos2;
+                fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
+            }
+        }
+    }
+    ff_set_fixed_vector(pulses, &fcb, 1.0, size);
+
+    /* Calculate gain for adaptive & fixed codebook signal.
+     * see ff_amr_set_fixed_gain(). */
+    idx = get_bits(gb, 7);
+    fcb_gain = expf(ff_dot_productf(s->gain_pred_err, gain_coeff, 6) -
+                    5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
+    acb_gain = wmavoice_gain_codebook_acb[idx];
+    pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
+                        -2.9957322736 /* log(0.05) */,
+                         1.6094379124 /* log(5.0)  */);
+
+    gain_weight = 8 >> frame_desc->log_n_blocks;
+    memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
+            sizeof(*s->gain_pred_err) * (6 - gain_weight));
+    for (n = 0; n < gain_weight; n++)
+        s->gain_pred_err[n] = pred_err;
+
+    /* Calculation of adaptive codebook */
+    if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
+        int len;
+        for (n = 0; n < size; n += len) {
+            int next_idx_sh16;
+            int abs_idx    = block_idx * size + n;
+            int pitch_sh16 = (s->last_pitch_val << 16) +
+                             s->pitch_diff_sh16 * abs_idx;
+            int pitch      = (pitch_sh16 + 0x6FFF) >> 16;
+            int idx_sh16   = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
+            idx            = idx_sh16 >> 16;
+            if (s->pitch_diff_sh16) {
+                if (s->pitch_diff_sh16 > 0) {
+                    next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
+                } else
+                    next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
+                len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
+                              1, size - n);
+            } else
+                len = size;
+
+            ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
+                                  wmavoice_ipol1_coeffs, 17,
+                                  idx, 9, len);
+        }
+    } else /* ACB_TYPE_HAMMING */ {
+        int block_pitch = block_pitch_sh2 >> 2;
+        idx             = block_pitch_sh2 & 3;
+        if (idx) {
+            ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
+                                  wmavoice_ipol2_coeffs, 4,
+                                  idx, 8, size);
+        } else
+            av_memcpy_backptr(excitation, sizeof(float) * block_pitch,
+                              sizeof(float) * size);
+    }
+
+    /* Interpolate ACB/FCB and use as excitation signal */
+    ff_weighted_vector_sumf(excitation, excitation, pulses,
+                            acb_gain, fcb_gain, size);
+}
+
+/**
+ * Parse data in a single block.
+ * @note we assume enough bits are available, caller should check.
+ *
+ * @param s WMA Voice decoding context private data
+ * @param gb bit I/O context
+ * @param block_idx index of the to-be-read block
+ * @param size amount of samples to be read in this block
+ * @param block_pitch_sh2 pitch for this block << 2
+ * @param lsps LSPs for (the end of) this frame
+ * @param prev_lsps LSPs for the last frame
+ * @param frame_desc frame type descriptor
+ * @param excitation target memory for the ACB+FCB interpolated signal
+ * @param synth target memory for the speech synthesis filter output
+ * @return 0 on success, <0 on error.
+ */
+static void synth_block(WMAVoiceContext *s, GetBitContext *gb,
+                        int block_idx, int size,
+                        int block_pitch_sh2,
+                        const double *lsps, const double *prev_lsps,
+                        const struct frame_type_desc *frame_desc,
+                        float *excitation, float *synth)
+{
+    double i_lsps[MAX_LSPS];
+    float lpcs[MAX_LSPS];
+    float fac;
+    int n;
+
+    if (frame_desc->acb_type == ACB_TYPE_NONE)
+        synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
+    else
+        synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
+                            frame_desc, excitation);
+
+    /* convert interpolated LSPs to LPCs */
+    fac = (block_idx + 0.5) / frame_desc->n_blocks;
+    for (n = 0; n < s->lsps; n++) // LSF -> LSP
+        i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
+    ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
+
+    /* Speech synthesis */
+    ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
+}
+
+/**
+ * Synthesize output samples for a single frame.
+ * @note we assume enough bits are available, caller should check.
+ *
+ * @param ctx WMA Voice decoder context
+ * @param gb bit I/O context (s->gb or one for cross-packet superframes)
+ * @param samples pointer to output sample buffer, has space for at least 160
+ *                samples
+ * @param lsps LSP array
+ * @param prev_lsps array of previous frame's LSPs
+ * @param excitation target buffer for excitation signal
+ * @param synth target buffer for synthesized speech data
+ * @return 0 on success, <0 on error.
+ */
+static int synth_frame(AVCodecContext *ctx, GetBitContext *gb,
+                       float *samples,
+                       const double *lsps, const double *prev_lsps,
+                       float *excitation, float *synth)
+{
+    WMAVoiceContext *s = ctx->priv_data;
+    int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val;
+    int pitch[MAX_BLOCKS], last_block_pitch;
+
+    /* Parse frame type ("frame header"), see frame_descs */
+    int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)],
+        block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
+
+    if (bd_idx < 0) {
+        av_log(ctx, AV_LOG_ERROR,
+               "Invalid frame type VLC code, skipping\n");
+        return -1;
+    }
+
+    /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
+    if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
+        /* Pitch is provided per frame, which is interpreted as the pitch of
+         * the last sample of the last block of this frame. We can interpolate
+         * the pitch of other blocks (and even pitch-per-sample) by gradually
+         * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
+        n_blocks_x2      = frame_descs[bd_idx].n_blocks << 1;
+        log_n_blocks_x2  = frame_descs[bd_idx].log_n_blocks + 1;
+        cur_pitch_val    = s->min_pitch_val + get_bits(gb, s->pitch_nbits);
+        cur_pitch_val    = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
+        if (s->last_acb_type == ACB_TYPE_NONE ||
+            20 * abs(cur_pitch_val - s->last_pitch_val) >
+                (cur_pitch_val + s->last_pitch_val))
+            s->last_pitch_val = cur_pitch_val;
+
+        /* pitch per block */
+        for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
+            int fac = n * 2 + 1;
+
+            pitch[n] = (MUL16(fac,                 cur_pitch_val) +
+                        MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
+                        frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
+        }
+
+        /* "pitch-diff-per-sample" for calculation of pitch per sample */
+        s->pitch_diff_sh16 =
+            ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE;
+    }
+
+    /* Global gain (if silence) and pitch-adaptive window coordinates */
+    switch (frame_descs[bd_idx].fcb_type) {
+    case FCB_TYPE_SILENCE:
+        s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];
+        break;
+    case FCB_TYPE_AW_PULSES:
+        aw_parse_coords(s, gb, pitch);
+        break;
+    }
+
+    for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
+        int bl_pitch_sh2;
+
+        /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
+        switch (frame_descs[bd_idx].acb_type) {
+        case ACB_TYPE_HAMMING: {
+            /* Pitch is given per block. Per-block pitches are encoded as an
+             * absolute value for the first block, and then delta values
+             * relative to this value) for all subsequent blocks. The scale of
+             * this pitch value is semi-logaritmic compared to its use in the
+             * decoder, so we convert it to normal scale also. */
+            int block_pitch,
+                t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
+                t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
+                t3 =  s->block_conv_table[3] - s->block_conv_table[2] + 1;
+
+            if (n == 0) {
+                block_pitch = get_bits(gb, s->block_pitch_nbits);
+            } else
+                block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
+                                 get_bits(gb, s->block_delta_pitch_nbits);
+            /* Convert last_ so that any next delta is within _range */
+            last_block_pitch = av_clip(block_pitch,
+                                       s->block_delta_pitch_hrange,
+                                       s->block_pitch_range -
+                                           s->block_delta_pitch_hrange);
+
+            /* Convert semi-log-style scale back to normal scale */
+            if (block_pitch < t1) {
+                bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
+            } else {
+                block_pitch -= t1;
+                if (block_pitch < t2) {
+                    bl_pitch_sh2 =
+                        (s->block_conv_table[1] << 2) + (block_pitch << 1);
+                } else {
+                    block_pitch -= t2;
+                    if (block_pitch < t3) {
+                        bl_pitch_sh2 =
+                            (s->block_conv_table[2] + block_pitch) << 2;
+                    } else
+                        bl_pitch_sh2 = s->block_conv_table[3] << 2;
+                }
+            }
+            pitch[n] = bl_pitch_sh2 >> 2;
+            break;
+        }
+
+        case ACB_TYPE_ASYMMETRIC: {
+            bl_pitch_sh2 = pitch[n] << 2;
+            break;
+        }
+
+        default: // ACB_TYPE_NONE has no pitch
+            bl_pitch_sh2 = 0;
+            break;
+        }
+
+        synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
+                    lsps, prev_lsps, &frame_descs[bd_idx],
+                    &excitation[n * block_nsamples],
+                    &synth[n * block_nsamples]);
+    }
+
+    /* Averaging projection filter, if applicable. Else, just copy samples
+     * from synthesis buffer */
+    if (s->do_apf) {
+        // FIXME this is where APF would take place, currently not implemented
+        av_log_missing_feature(ctx, "APF", 0);
+        s->do_apf = 0;
+    } //else
+        for (n = 0; n < 160; n++)
+            samples[n] = av_clipf(synth[n], -1.0, 1.0);
+
+    /* Cache values for next frame */
+    s->frame_cntr++;
+    if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
+    s->last_acb_type = frame_descs[bd_idx].acb_type;
+    switch (frame_descs[bd_idx].acb_type) {
+    case ACB_TYPE_NONE:
+        s->last_pitch_val = 0;
+        break;
+    case ACB_TYPE_ASYMMETRIC:
+        s->last_pitch_val = cur_pitch_val;
+        break;
+    case ACB_TYPE_HAMMING:
+        s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
+        break;
+    }
+
+    return 0;
+}
+
+/**
+ * Ensure minimum value for first item, maximum value for last value,
+ * proper spacing between each value and proper ordering.
+ *
+ * @param lsps array of LSPs
+ * @param num size of LSP array
+ *
+ * @note basically a double version of #ff_acelp_reorder_lsf(), might be
+ *       useful to put in a generic location later on. Parts are also
+ *       present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
+ *       which is in float.
+ */
+static void stabilize_lsps(double *lsps, int num)
+{
+    int n, m, l;
+
+    /* set minimum value for first, maximum value for last and minimum
+     * spacing between LSF values.
+     * Very similar to ff_set_min_dist_lsf(), but in double. */
+    lsps[0]       = FFMAX(lsps[0],       0.0015 * M_PI);
+    for (n = 1; n < num; n++)
+        lsps[n]   = FFMAX(lsps[n],       lsps[n - 1] + 0.0125 * M_PI);
+    lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
+
+    /* reorder (looks like one-time / non-recursed bubblesort).
+     * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
+    for (n = 1; n < num; n++) {
+        if (lsps[n] < lsps[n - 1]) {
+            for (m = 1; m < num; m++) {
+                double tmp = lsps[m];
+                for (l = m - 1; l >= 0; l--) {
+                    if (lsps[l] <= tmp) break;
+                    lsps[l + 1] = lsps[l];
+                }
+                lsps[l + 1] = tmp;
+            }
+            break;
+        }
+    }
+}
+
+/**
+ * Test if there's enough bits to read 1 superframe.
+ *
+ * @param orig_gb bit I/O context used for reading. This function
+ *                does not modify the state of the bitreader; it
+ *                only uses it to copy the current stream position
+ * @param s WMA Voice decoding context private data
+ * @returns -1 if unsupported, 1 on not enough bits or 0 if OK.
+ */
+static int check_bits_for_superframe(GetBitContext *orig_gb,
+                                     WMAVoiceContext *s)
+{
+    GetBitContext s_gb, *gb = &s_gb;
+    int n, need_bits, bd_idx;
+    const struct frame_type_desc *frame_desc;
+
+    /* initialize a copy */
+    init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits);
+    skip_bits_long(gb, get_bits_count(orig_gb));
+    assert(get_bits_left(gb) == get_bits_left(orig_gb));
+
+    /* superframe header */
+    if (get_bits_left(gb) < 14)
+        return 1;
+    if (!get_bits1(gb))
+        return -1;                        // WMAPro-in-WMAVoice superframe
+    if (get_bits1(gb)) skip_bits(gb, 12); // number of  samples in superframe
+    if (s->has_residual_lsps) {           // residual LSPs (for all frames)
+        if (get_bits_left(gb) < s->sframe_lsp_bitsize)
+            return 1;
+        skip_bits_long(gb, s->sframe_lsp_bitsize);
+    }
+
+    /* frames */
+    for (n = 0; n < MAX_FRAMES; n++) {
+        int aw_idx_is_ext = 0;
+
+        if (!s->has_residual_lsps) {     // independent LSPs (per-frame)
+           if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1;
+           skip_bits_long(gb, s->frame_lsp_bitsize);
+        }
+        bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)];
+        if (bd_idx < 0)
+            return -1;                   // invalid frame type VLC code
+        frame_desc = &frame_descs[bd_idx];
+        if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
+            if (get_bits_left(gb) < s->pitch_nbits)
+                return 1;
+            skip_bits_long(gb, s->pitch_nbits);
+        }
+        if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
+            skip_bits(gb, 8);
+        } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
+            int tmp = get_bits(gb, 6);
+            if (tmp >= 0x36) {
+                skip_bits(gb, 2);
+                aw_idx_is_ext = 1;
+            }
+        }
+
+        /* blocks */
+        if (frame_desc->acb_type == ACB_TYPE_HAMMING) {
+            need_bits = s->block_pitch_nbits +
+                (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits;
+        } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
+            need_bits = 2 * !aw_idx_is_ext;
+        } else
+            need_bits = 0;
+        need_bits += frame_desc->frame_size;
+        if (get_bits_left(gb) < need_bits)
+            return 1;
+        skip_bits_long(gb, need_bits);
+    }
+
+    return 0;
+}
+
+/**
+ * Synthesize output samples for a single superframe. If we have any data
+ * cached in s->sframe_cache, that will be used instead of whatever is loaded
+ * in s->gb.
+ *
+ * WMA Voice superframes contain 3 frames, each containing 160 audio samples,
+ * to give a total of 480 samples per frame. See #synth_frame() for frame
+ * parsing. In addition to 3 frames, superframes can also contain the LSPs
+ * (if these are globally specified for all frames (residually); they can
+ * also be specified individually per-frame. See the s->has_residual_lsps
+ * option), and can specify the number of samples encoded in this superframe
+ * (if less than 480), usually used to prevent blanks at track boundaries.
+ *
+ * @param ctx WMA Voice decoder context
+ * @param samples pointer to output buffer for voice samples
+ * @param data_size pointer containing the size of #samples on input, and the
+ *                  amount of #samples filled on output
+ * @return 0 on success, <0 on error or 1 if there was not enough data to
+ *         fully parse the superframe
+ */
+static int synth_superframe(AVCodecContext *ctx,
+                            float *samples, int *data_size)
+{
+    WMAVoiceContext *s = ctx->priv_data;
+    GetBitContext *gb = &s->gb, s_gb;
+    int n, res, n_samples = 480;
+    double lsps[MAX_FRAMES][MAX_LSPS];
+    const double *mean_lsf = s->lsps == 16 ?
+        wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
+    float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
+    float synth[MAX_LSPS + MAX_SFRAMESIZE];
+
+    memcpy(synth,      s->synth_history,
+           s->lsps             * sizeof(*synth));
+    memcpy(excitation, s->excitation_history,
+           s->history_nsamples * sizeof(*excitation));
+
+    if (s->sframe_cache_size > 0) {
+        gb = &s_gb;
+        init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
+        s->sframe_cache_size = 0;
+    }
+
+    if ((res = check_bits_for_superframe(gb, s)) == 1) return 1;
+
+    /* First bit is speech/music bit, it differentiates between WMAVoice
+     * speech samples (the actual codec) and WMAVoice music samples, which
+     * are really WMAPro-in-WMAVoice-superframes. I've never seen those in
+     * the wild yet. */
+    if (!get_bits1(gb)) {
+        av_log_missing_feature(ctx, "WMAPro-in-WMAVoice support", 1);
+        return -1;
+    }
+
+    /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */
+    if (get_bits1(gb)) {
+        if ((n_samples = get_bits(gb, 12)) > 480) {
+            av_log(ctx, AV_LOG_ERROR,
+                   "Superframe encodes >480 samples (%d), not allowed\n",
+                   n_samples);
+            return -1;
+        }
+    }
+    /* Parse LSPs, if global for the superframe (can also be per-frame). */
+    if (s->has_residual_lsps) {
+        double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2];
+
+        for (n = 0; n < s->lsps; n++)
+            prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
+
+        if (s->lsps == 10) {
+            dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
+        } else /* s->lsps == 16 */
+            dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
+
+        for (n = 0; n < s->lsps; n++) {
+            lsps[0][n]  = mean_lsf[n] + (a1[n]           - a2[n * 2]);
+            lsps[1][n]  = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]);
+            lsps[2][n] += mean_lsf[n];
+        }
+        for (n = 0; n < 3; n++)
+            stabilize_lsps(lsps[n], s->lsps);
+    }
+
+    /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */
+    for (n = 0; n < 3; n++) {
+        if (!s->has_residual_lsps) {
+            int m;
+
+            if (s->lsps == 10) {
+                dequant_lsp10i(gb, lsps[n]);
+            } else /* s->lsps == 16 */
+                dequant_lsp16i(gb, lsps[n]);
+
+            for (m = 0; m < s->lsps; m++)
+                lsps[n][m] += mean_lsf[m];
+            stabilize_lsps(lsps[n], s->lsps);
+        }
+
+        if ((res = synth_frame(ctx, gb,
+                               &samples[n * MAX_FRAMESIZE],
+                               lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
+                               &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
+                               &synth[s->lsps + n * MAX_FRAMESIZE])))
+            return res;
+    }
+
+    /* Statistics? FIXME - we don't check for length, a slight overrun
+     * will be caught by internal buffer padding, and anything else
+     * will be skipped, not read. */
+    if (get_bits1(gb)) {
+        res = get_bits(gb, 4);
+        skip_bits(gb, 10 * (res + 1));
+    }
+
+    /* Specify nr. of output samples */
+    *data_size = n_samples * sizeof(float);
+
+    /* Update history */
+    memcpy(s->prev_lsps,           lsps[2],
+           s->lsps             * sizeof(*s->prev_lsps));
+    memcpy(s->synth_history,      &synth[MAX_SFRAMESIZE],
+           s->lsps             * sizeof(*synth));
+    memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
+           s->history_nsamples * sizeof(*excitation));
+
+    return 0;
+}
+
+/**
+ * Parse the packet header at the start of each packet (input data to this
+ * decoder).
+ *
+ * @param s WMA Voice decoding context private data
+ * @returns 1 if not enough bits were available, or 0 on success.
+ */
+static int parse_packet_header(WMAVoiceContext *s)
+{
+    GetBitContext *gb = &s->gb;
+    unsigned int res;
+
+    if (get_bits_left(gb) < 11)
+        return 1;
+    skip_bits(gb, 4);          // packet sequence number
+    s->has_residual_lsps = get_bits1(gb);
+    do {
+        res = get_bits(gb, 6); // number of superframes per packet
+                               // (minus first one if there is spillover)
+        if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize)
+            return 1;
+    } while (res == 0x3F);
+    s->spillover_nbits   = get_bits(gb, s->spillover_bitsize);
+
+    return 0;
+}
+
+/**
+ * Copy (unaligned) bits from gb/data/size to pb.
+ *
+ * @param pb target buffer to copy bits into
+ * @param data source buffer to copy bits from
+ * @param size size of the source data, in bytes
+ * @param gb bit I/O context specifying the current position in the source.
+ *           data. This function might use this to align the bit position to
+ *           a whole-byte boundary before calling #ff_copy_bits() on aligned
+ *           source data
+ * @param nbits the amount of bits to copy from source to target
+ *
+ * @note after calling this function, the current position in the input bit
+ *       I/O context is undefined.
+ */
+static void copy_bits(PutBitContext *pb,
+                      const uint8_t *data, int size,
+                      GetBitContext *gb, int nbits)
+{
+    int rmn_bytes, rmn_bits;
+
+    rmn_bits = rmn_bytes = get_bits_left(gb);
+    if (rmn_bits < nbits)
+        return;
+    rmn_bits &= 7; rmn_bytes >>= 3;
+    if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
+        put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
+    ff_copy_bits(pb, data + size - rmn_bytes,
+                 FFMIN(nbits - rmn_bits, rmn_bytes << 3));
+}
+
+/**
+ * Packet decoding: a packet is anything that the (ASF) demuxer contains,
+ * and we expect that the demuxer / application provides it to us as such
+ * (else you'll probably get garbage as output). Every packet has a size of
+ * ctx->block_align bytes, starts with a packet header (see
+ * #parse_packet_header()), and then a series of superframes. Superframe
+ * boundaries may exceed packets, i.e. superframes can split data over
+ * multiple (two) packets.
+ *
+ * For more information about frames, see #synth_superframe().
+ */
+static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
+                                  int *data_size, AVPacket *avpkt)
+{
+    WMAVoiceContext *s = ctx->priv_data;
+    GetBitContext *gb = &s->gb;
+    int size, res, pos;
+
+    if (*data_size < 480 * sizeof(float)) {
+        av_log(ctx, AV_LOG_ERROR,
+               "Output buffer too small (%d given - %lu needed)\n",
+               *data_size, 480 * sizeof(float));
+        return -1;
+    }
+    *data_size = 0;
+
+    /* Packets are sometimes a multiple of ctx->block_align, with a packet
+     * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
+     * feeds us ASF packets, which may concatenate multiple "codec" packets
+     * in a single "muxer" packet, so we artificially emulate that by
+     * capping the packet size at ctx->block_align. */
+    for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
+    if (!size)
+        return 0;
+    init_get_bits(&s->gb, avpkt->data, size << 3);
+
+    /* size == ctx->block_align is used to indicate whether we are dealing with
+     * a new packet or a packet of which we already read the packet header
+     * previously. */
+    if (size == ctx->block_align) { // new packet header
+        if ((res = parse_packet_header(s)) < 0)
+            return res;
+
+        /* If the packet header specifies a s->spillover_nbits, then we want
+         * to push out all data of the previous packet (+ spillover) before
+         * continuing to parse new superframes in the current packet. */
+        if (s->spillover_nbits > 0) {
+            if (s->sframe_cache_size > 0) {
+                int cnt = get_bits_count(gb);
+                copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
+                flush_put_bits(&s->pb);
+                s->sframe_cache_size += s->spillover_nbits;
+                if ((res = synth_superframe(ctx, data, data_size)) == 0 &&
+                    *data_size > 0) {
+                    cnt += s->spillover_nbits;
+                    s->skip_bits_next = cnt & 7;
+                    return cnt >> 3;
+                } else
+                    skip_bits_long (gb, s->spillover_nbits - cnt +
+                                    get_bits_count(gb)); // resync
+            } else
+                skip_bits_long(gb, s->spillover_nbits);  // resync
+        }
+    } else if (s->skip_bits_next)
+        skip_bits(gb, s->skip_bits_next);
+
+    /* Try parsing superframes in current packet */
+    s->sframe_cache_size = 0;
+    s->skip_bits_next = 0;
+    pos = get_bits_left(gb);
+    if ((res = synth_superframe(ctx, data, data_size)) < 0) {
+        return res;
+    } else if (*data_size > 0) {
+        int cnt = get_bits_count(gb);
+        s->skip_bits_next = cnt & 7;
+        return cnt >> 3;
+    } else if ((s->sframe_cache_size = pos) > 0) {
+        /* rewind bit reader to start of last (incomplete) superframe... */
+        init_get_bits(gb, avpkt->data, size << 3);
+        skip_bits_long(gb, (size << 3) - pos);
+        assert(get_bits_left(gb) == pos);
+
+        /* ...and cache it for spillover in next packet */
+        init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
+        copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size);
+        // FIXME bad - just copy bytes as whole and add use the
+        // skip_bits_next field
+    }
+
+    return size;
+}
+
+static av_cold void wmavoice_flush(AVCodecContext *ctx)
+{
+    WMAVoiceContext *s = ctx->priv_data;
+    int n;
+
+    s->sframe_cache_size = 0;
+    s->skip_bits_next    = 0;
+    for (n = 0; n < s->lsps; n++)
+        s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
+    memset(s->excitation_history, 0,
+           sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
+    memset(s->synth_history,      0,
+           sizeof(*s->synth_history)      * MAX_LSPS);
+    memset(s->gain_pred_err,      0,
+           sizeof(s->gain_pred_err));
+}
+
+AVCodec wmavoice_decoder = {
+    "wmavoice",
+    CODEC_TYPE_AUDIO,
+    CODEC_ID_WMAVOICE,
+    sizeof(WMAVoiceContext),
+    wmavoice_decode_init,
+    NULL,
+    NULL,
+    wmavoice_decode_packet,
+    CODEC_CAP_SUBFRAMES,
+    .flush     = wmavoice_flush,
+    .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
+};